similar to: RE: how to manage Digium TDM04B outgoing calls

Is anyone out there using an "AudioCodes MP108 8-Port FXO Analog Gateway (SIP)" with asterisk to support both inbound and outbound calling? If so, I'm interested to hear how it works, and I'd love to see some example confs (both in sip.conf and on the MP108). This product has been recommended to me by a SNOM/Asterisk-friendly distributor, but I would like a second opinion

Hello all, Has anybody cofigured in SIP the Audiocodes MP108 FXS in a way that each port is an extension of the Asterisk Box ? So each port can have it's own mailbox, etc ? Regards, Jorge A.

I'm installing my first Asterisk server. I have a TDM04B card installed in my asterisk server (4x FXO ports). I have 5 Cisco IP phone 7960 working fine on asterisk using SIP. My configuration to receive call is working as expected meaning anyone calling on one of the 4 FXO ports is answer by asterisk and asked to enter the extension of the person to reach and then it is transfer on the

-----Original Message----- My question concern outgoing calls. How can I configure my extensions.conf to get a PSTN line on my TDM04B card in the following order : first trying on the channel 4 then if 4 is busy then switch to 3 if 3 is busy then switch to 2 and if 2 is busy then say there's no more line available. I don't want to dial on the first channel as it's my main number

Anyone tried the Clipcomm CG-410? I'm tired of beeing unable to get rid correctly of the echo problem. I have 3 TDM04B installed in one server. It was working fine when I had only one card installed in the server but since I installed 2 more then I can't get rid of it what ever I try... So now I'm looking at other solutions. And I saw that Clipcomm makes a router with 4 FXO ports

Hello. Has someone been able to make work faxes over sip, i have one mp108 fxo and one mp108 fxs, my setup is : telco analog line -----> mp108fxo -----> Asterisk ------> mp108fxs -------> fax machine 1) Asterisk detects the tone from the sending fax ( i am receiving ) but looks for extension 'ff' not 'fax', ( at least that's what * complaint, invalid

Hello, I have received yet another new phone today, the ClipComm 101 (http://www.clipcomm.co.kr/eng/e_product/e_product_voip_ip_phone.html) I bought it for $165 directly from the Korean Manufacturer(No US distributer yet). Here are the features: - Built-in NAT functionality, you can switch from Hub to Nat, great for home DSL/Cable users - This includes some limited port forwarding

There are somes products available claiming to connect a BT headset, a cell phone and a phone land line all together. I've found some : http://www.geekzone.co.nz/content.asp?contentid=2079 http://www.clipcomm.co.kr/ The clipcomm BS-A101 sample price is : $570. It's VoIP land phone that can make/receive a voice call using your CTP cell phone or BT-enabled headset. Maybe the best system

I have searched for how to locate echo cancelation on SIP clients, but cant find anything and echocancel=y doesnt seem to have any effect. Configuration: CVS-HEAD from last month iPAQ h5500 with SJPhone (gsm/ulaw/alaw) Problem description: When I place or receive a call I hear a faint delayed echo of myself. The other party hears a really bad nonmuted echo that makes the call unusable. Aside

I have installed Digium TDM04B with the latest CVS. However I have encountered following problems: 1. When it dials out, many times the digits are not properly recognized by telco as I hear the announcement "please check the number and dial again" although I see on the screen that the dialed number is correct. 2. When the call is forwarded outside, with something like exten =>

Hello, I'm interested in implementing DID functionality with the Digium TDM04B adapter. Is DID supported with POTS? Are there any caveeats or drawbacks that I should be aware before proceeding? This pbx will be implemented in eastern europe. Thanks in advance. -- Martin Spasov <mspasov@techno-link.com>

Hi All, I'd really like to see a show of hands with regard to people's experience with FXO interfaces. I own a few X100p cards and have had nothing but problems with them. I also took part in Sipura's beta program, for the SPA-3000. While it can be an improvement over the X100p, it presently has echo problems that make it unusable. Sipura has not acknowledged the problem ( at least

I already have one Digium TDM04B installed in my server working fine. I just received 2 more so I did added them to my server. When I booted again linux told me that it found new hardware I said ignore. Then I log in as usual. I did ztcfg -vv to see if it sees 12 channels now instead of 4 but it only see 4. So then I did again modprobe zaptel, then modprobe wcfxo and wcfxs hoping it would see

I am currently installing an Audiocodes MP-108 with Asterisk in Singapore. I am able to make outgoing calls but incomming calls just get cut by the MP-108 I can see the corrosponding frontpanel led light p but the call gets cut so I'm very confused about what could be wrong. Could this problem have something to do

Hi, A week or so ago I wrote about the problems I was having using a Digium TDM04B card in a Dell PowerEdge 750 IU running Fedora Core 1. Digium steadfastly indicates their card won't work in any PowerEdge 650, 700 and 750 series machines "do (sic) to a failure in interrupt handling between the pci bus and the card." My question to the list is: for those of you using a TDM04B

Hi pros, Newbie to asterisk, need some help. My existing senerio is we have 6 analog quintums and 1 digital H323, and our gatekeeper is gnugk openh323 located in US. Our business is Call Center and our method of dial is using prefix and gateway IP provided my Carrier. I also brought two AudioCodes MP108 8 FXS gateways, as our gateway runs on h323 my friend suggested to go for Asterisk. If

Does anyone have any success using AudioCodes FXO terminating calls ? Ehsan -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20051002/5cfef736/attachment.htm

Hi, I'm experiencing some strange problems with out Digium card. First the details abount hardware and software: Digium, Inc. Wildcard B410 quad-BRI card (rev 01) Asterisk 1.6.0.20 dahdi-linux-complete-2.2.1-rc2+2.2.1-rc2 libpri-1.4.10.2.tar.gz The problem now is that there are a number of clients that I call and suddenly the connection drops. Just a few seconds later the clients calls

I would like to install a TDM04B with Asterisk@Home 2.4 and Asterisk@Home 1.5 but I didn't find documentation about this installation. Thanks in advance, Nelson -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060206/85ff5252/attachment.htm

Hello, I have installed asterisk version 1.2.12 and latest zaptel modules. but i can see some IRQ conflicts on the server. iam uisng two TDM04B cards. according to my previous knowledge on asterisk verison 1.07 asterisk has given lot of erros when starting if you have assigned the same IRQ number to any other device. My question is new releae version 1.2.12 has resolved the IRQ issue ?? TDM04B